P18-6 Sampling Rate Discrimination: 44.1 kHz vs. 88.2 kHz—Amandine Pras, Catherine Guastavino, McGill University - Montreal, Quebec, CanadaIt is currently common practice for sound engineers to record digital music using high-resolution formats, and then down sample the files to 44.1 kHz for commercial release. This study aims at investigating whether listeners can perceive differences between musical files recorded at 44.1 kHz and 88.2 kHz with the same analog chain and type of AD-converter. Sixteen expert listeners were asked to compare 3 versions (44.1 kHz, 88.2 kHz, and the 88.2 kHz version down-sampled to 44.1 kHz) of 5 musical excerpts in a blind ABX task. Overall, participants were able to discriminate between files recorded at 88.2 kHz and their 44.1 kHz down-sampled version. Furthermore, for the orchestral excerpt, they were able to discriminate between files recorded at 88.2 kHz and files recorded at 44.1 kHz. Convention Paper 8101

I read the full paper and I think there may be some transitive errors here.

It looks like in some cases listeners can detect a difference between 88.1 and 44.1 native , not between 44.1 down and 44.1 native and not between 88.1 and 44.1 down for the same material. This implies that 44.1 native lacks something but 88.1 to 44.1 retains what was lost in the 44.1 native. This does not appear to make sense since both 44.1 native and 44.1 down have the same limits (give or take dithering concerns) - I am confused by this. ???

This does not appear to make sense since both 44.1 native and 44.1 down have the same limits (give or take dithering concerns) - I am confused by this. ???

If the statistics were significant, it would make sense, because downsampling 88.2 to 44.1, you can use perfect digital antialias filters, while recording directly at 44.1, you have to use analog antialias filters, in order for your signal to be lowpassed before it reaches the ADC.

This does not appear to make sense since both 44.1 native and 44.1 down have the same limits (give or take dithering concerns) - I am confused by this. ???

If the statistics were significant, it would make sense, because downsampling 88.2 to 44.1, you can use perfect digital antialias filters, while recording directly at 44.1, you have to use analog antialias filters, in order for your signal to be lowpassed before it reaches the ADC.

Modern ADCs do have analog anti-aliasing filters, but they are relatively simple and operate at ultrasonic frequencies. The brick wall that is right up against the audio band is digital and therefore the overall performance can be very similar to what you get if you record at a higher sample rate and downsample in the digital domain. Note that there can be considerable techncal variation in the details of how the digital filtering is implemented, whether in the ADC or applied later on.

Modern ADCs do have analog anti-aliasing filters, but they are relatively simple and operate at ultrasonic frequencies. The brick wall that is right up against the audio band is digital and therefore the overall performance can be very similar to what you get if you record at a higher sample rate and downsample in the digital domain. Note that there can be considerable techncal variation in the details of how the digital filtering is implemented, whether in the ADC or applied later on.

The part about the final filtering being digital seems right, as far as I understand from reading, but based on my experiments, and those of a few others, the result of recording at 44.1 is never like that from recording at a 88.2 or 96 and downsampling with good software, as I pointed out earlier in this thread and in at least two others in HA (based on results using test tones, the only way to actually observe the final product). Do you have evidence that some soundcards really do better?

The part about the final filtering being digital seems right, as far as I understand from reading, but based on my experiments, and those of a few others, the result of recording at 44.1 is never like that from recording at a 88.2 or 96 and downsampling with good software, as I pointed out earlier in this thread and in at least two others in HA (based on results using test tones, the only way to actually observe the final product). Do you have evidence that some soundcards really do better?

I agree Andy - I've never seen an A>D (or D>A) that comes close to achieving the kind of truly brick wall filtering that you get in Cool Edit Pro's resampling.

Whether this matters at all is another question, but it's an easily measurable difference.

I take Arny's point that they're all different (+ many are programmable), but none seem to make the effort to include a several thousand tap FIR filter.

I take Arny's point that they're all different (+ many are programmable), but none seem to make the effort to include a several thousand tap FIR filter.

That's because DACs are usually designed by engineers with some notion of costs and benefits. Some of the best converter chips seem to sell for under $7 (single unit!) and that's still too low to allow including a 2 GHz processor on the same chip.

I can still remember when a first rate converter chip cost more than $20!